1. Field of the Invention
The present invention relates to a system and method for changing codec information to provide a VoIP terminal with a coloring service, and more specifically to a system and method for changing codec information to provide a VoIP terminal with a coloring service, the VoIP terminal providing a more efficient coloring service by applying different compression schemes depending on a calling state and a call connecting state between VoIP terminals.
2. Description of the Related Art
Voice over Internet Protocol (VoIP) is a transmission scheme to transmit voice information using an IP. That is, VoIP does not use a traditional protocol based on a line like an existing Public Switched Telephone Network (PSTN), but rather transmits voice information in digital form within discrete packets. By implementing an integrated telephone service utilizing the existing IP network as is with VoIP and Internet telephone technologies, telephone users can use toll and international telephone services in Internet and Intranet environments with only a local telephone charge.
The most basic function to make a synergy effect by combining voice and data in VoIP technology is a voice service function using an IP. Particularly, representative codec schemes to compress voice transferred between terminals upon voice communication using VoIP are G.711, G.729, G723.1 and G.726. However, the algorithmic Mean Option Score (MOS) deterioration as shown in Table 1 below occurs depending on a type of coding scheme with which the voice is transmitted.
TABLE 1CompressionProcessing 1Frame sizeschemeBit rate (kbps)(mips)(unit)MOS scoreG.711PCM640.340.1254.1G.726PCM32140.1253.85G.72816330.6253.61LD-CELPG.729820103.92CS-ACELPG.729 × 2820103.27EncodingG.729 × 3820102.68EncodingG.729a810.5103.7CS-ACELPG.723.16.316303.9MP-MLQG.723.15.316303.65ACELP
That is, a voice quality deterioration phenomenon occurs due to a characteristic of a codec technology used in VoIP as shown in Table 1. Accordingly, in order to provide a voice quality service on VoIP which is similar to that of PSTN, the G.711 PCM scheme should be utilized, where algorithmic MOS deterioration is the minimum. However, since the G.711 PCM scheme wastes more bandwidth compared with other codec schemes, the MOS of Table 1 should not be solely considered in diagram of the bandwidth of a network. Particularly, although the G.723.1 MP-MLQ scheme and G.729 CS-ACELP scheme have MOS values of 3.9 and 3.92 that are lower than that of G.711 PCM scheme, they have better MOS performance than that of the G.711 PCM scheme in contrast with the bandwidth since the MOS deterioration is hard to perceive compared with the G.711 PCM scheme. Therefore, a codec technology that is generally used in a current VoIP gateway is the G.729 CS-ACELP scheme that has an excellent MOS value compared to the bandwidth.
Communication networks, in which the call charge is a main income source, have recently tried to increase profits by adding various additional services, and accordingly, service provisions using a tone generator have increased. A representative service includes a coloring service such as a Coloring and a Ring To Me, which is provided before a speech path has been established. This is a CS-ACELP scheme based on a narrowband voice conversion and voice property, which causes consumer dissatisfaction with the speech quality unlike the voice service, such that the user has a bad impression of VoIP service. In order to overcome this problem, VoIP service can be provided by utilizing the G.711 PCM scheme which has a high MOS. However, it cannot be a desirable solution in that it causes a problem in using network resources efficiently and can create a limited inconvenience for a data service due to performance of a bandwidth resource reservation function for guaranteeing the speech quality of VoIP.
In a call setup process using H.323 protocol in a VoIP system, a sending Media Gateway (MG) transmits “Q.931 Set up” message to a receiving MG. That is, the sending MG transmits a list of several codec information to the receiving MG so as to negotiate with the receiving MG. Accordingly, the receiving MG transmits a confirmation message of “Q.931 Call Proceeding” with respect to the reception of the “Q.931 Set up” message to the sending MG, and selects a codec scheme set by the receiving MG from the codec information list which is transmitted from the sending MG and then transmits the “Q.931 Alerting” message to the sending MG 1. Accordingly, in a state where a ring back tone is transmitted from the receiving MG to the sending MG through a Real-time Transport Protocol (RTP), when a hook off event occurs and a voice call connection is performed, the “Q.931 Connect” message is transmitted from the receiving MG to the sending MG so that a voice communication is performed through the RTP.
In a call setup process using an Session Initiation Protocol (SIP) of a VoIP system, a sending MG transmits a speech request message of “INVITE” to a receiving MG. That is, the sending MG transmits a list of several codec information to the receiving MG so as to negotiate with the receiving MG. Accordingly, the receiving MG transmits a confirmation message of “100 Trying” with respect to the reception of the “INVITE” message to the sending MG, and selects a codec scheme set by the receiving MG from the codec information list which is transmitted from the sending MG and then transmits the “180 Ringing” message to the sending MG. Accordingly, in the state where a ring back tone is transmitted from the receiving MG to the sending MG through an RTP, when a receiving terminal user hooks off to make a voice call connection, “200 OK” message is transmitted from the receiving MG to the sending MG and a confirmation message of “ACK” with respect to the transmission is transmitted to the receiving MG, so that a voice communication is performed through an RTP.
As such, in most current VoIP gateways, a determination of a codec scheme for the VoIP service is made at a time point of transmitting an alerting message (Q.931 alerting message in the case of H.323 and 180 response message in the case of SIP). The set codec scheme information remains unchanged even after the receiving terminal has been hooked off. Accordingly, since the same codec as that in the ring-back tone service is used even after the receiving terminal is hooked off, it is confronted by a trade-off problem between the above mentioned MOS value and the bandwidth. That is, in utilizing the G.711 scheme that is a codec of PCM scheme for the sake of qualitative provision of a coloring service, a drawback occurs in that the bandwidth is wasted compared with the G.723.1 and G.729 schemes. In the G.723.1 and G.729 schemes having excellent bandwidth performance, a drawback occurs in that the qualitative service for the coloring service cannot be provided due to the voice property based algorithm. When various QoS algorithms to guarantee the speech quality are used in a network where voice and data are simultaneously serviced, a drawback occurs in the data service due to the relative waste of bandwidth. Accordingly, when a tone generator service, such as the coloring and RING TO ME is used, which is not a simple ring-back tone but has recently been highlighted as an effective additional service in voice communication, it is difficult to provide the qualitative service similar to the PSTN even with the G.723.1 and G.729 schemes whose technical origin is based on voice properties.